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<H1>Full command line switch reference</H1>
<P> <font size="-1">note: Options which could exist without being documented 
  here are considered as experimental ones. Such experimental options should usually 
  not be used.</font> 
<P> 
<TABLE CELLPADDING=3 BORDER="1">
  <TR VALIGN="TOP"> 
    <TD ALIGN="LEFT" nowrap><b>switch</b></TD>
    <TD ALIGN="LEFT" nowrap><b>parameter</b></TD>
  </TR>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#a">-a</a></kbd></td>
    <td align="LEFT" nowrap>downmix stereo file to mono</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-abr">--abr</a></kbd></td>
    <td align="LEFT" nowrap>average bitrate encoding</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-allshort">--allshort</a></kbd></td>
    <td align="LEFT" nowrap>use short blocks only</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-athlower">--athlower</a></kbd></td>
    <td align="LEFT" nowrap>lower the ATH</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-athonly">--athonly</a></kbd></td>
    <td align="LEFT" nowrap>ATH only</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-athshort">--athshort</a></kbd></td>
    <td align="LEFT" nowrap>ATH only for short blocks</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-athtype">--athtype</a></kbd></td>
    <td align="LEFT" nowrap>select ATH type</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#b">-b</a></kbd></td>
    <td align="LEFT" nowrap>bitrate (8...320)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#Bmax">-B</a></kbd></td>
    <td align="LEFT" nowrap>max VBR/ABR bitrate (8...320)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-bitwidth">--bitwidth</a></kbd></td>
    <td align="LEFT" nowrap>input bit width</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#c">-c</a></kbd></td>
    <td align="LEFT" nowrap>copyright</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-cbr">--cbr</a></kbd></td>
    <td align="LEFT" nowrap>enforce use of constant bitrate</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-clipdetect">--clipdetect</a></kbd></td>
    <td align="LEFT" nowrap>clipping detection</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-comp">--comp</a></kbd></td>
    <td align="LEFT" nowrap>choose compression ratio</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-cwlimit">--cwlimit</a></kbd></td>
    <td align="LEFT" nowrap>tonality limit</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#d">-d</a></kbd></td>
    <td align="LEFT" nowrap>block type control</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-decode">--decode</a></kbd></td>
    <td align="LEFT" nowrap>decoding only</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-disptime">--disptime</a></kbd></td>
    <td align="LEFT" nowrap>time between display updates</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#e">-e</a></kbd></td>
    <td align="LEFT" nowrap>de-emphasis (<b>n</b>, 5, c)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#f">-f</a></kbd></td>
    <td align="LEFT" nowrap> fast mode</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#FF">-F</a></kbd></td>
    <td align="LEFT" nowrap> strictly enforce the -b option</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-freeformat">--freeformat</a></kbd></td>
    <td align="LEFT" nowrap> free format bitstream</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#h">-h</a></kbd></td>
    <td align="LEFT" nowrap>high quality</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-help">--help</a></kbd></td>
    <td align="LEFT" nowrap> help</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass</a></kbd></td>
    <td align="LEFT" nowrap> highpass filtering frequency in kHz</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass-width</a></kbd></td>
    <td align="LEFT" nowrap> width of highpass filtering in kHz</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#k">-k</a></kbd></td>
    <td align="LEFT" nowrap> full bandwidth</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-lowpass">--lowpass</a></kbd></td>
    <td align="LEFT" nowrap> lowpass filtering frequency in kHz</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-lowpass-width">--lowpass-width</a></kbd></td>
    <td align="LEFT" nowrap> width of lowpass filtering in kHz</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#m">-m</a></kbd></td>
    <td align="LEFT" nowrap>stereo mode (s, <b>j</b>, f, m)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-mp1input">--mp1input</a></kbd></td>
    <td align="LEFT" nowrap>MPEG Layer I input file</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-mp2input">--mp2input</a></kbd></td>
    <td align="LEFT" nowrap>MPEG Layer II input file</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-mp3input">--mp3input</a></kbd></td>
    <td align="LEFT" nowrap>MPEG Layer III input file</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-noath">--noath</a></kbd></td>
    <td align="LEFT" nowrap>disable ATH</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-noasm">--noasm</a></kbd></td>
    <td align="LEFT" nowrap>disable assembly optimizations (mmx/3dnow/sse)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-nohist">--nohist</a></kbd></td>
    <td align="LEFT" nowrap>disable histogram display</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-noreplaygain">--noreplaygain</a></kbd></td>
    <td align="LEFT" nowrap>disable ReplayGain analysis</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-nores">--nores</a></kbd></td>
    <td align="LEFT" nowrap>disable bit reservoir</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-noshort">--noshort</a></kbd></td>
    <td align="LEFT" nowrap>disable short blocks frames</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-notemp">--notemp</a></kbd></td>
    <td align="LEFT" nowrap>disable temporal masking</td>
  </tr>
  <TR VALIGN="TOP"> 
    <TD ALIGN="LEFT" nowrap><kbd><a href="#o">-o</a></kbd></TD>
    <TD ALIGN="LEFT" nowrap>non-original</TD>
  </TR>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#p">-p</a></kbd></td>
    <td align="LEFT" nowrap>error protection</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-preset">--preset</a></kbd></td>
    <td align="LEFT" nowrap>use built-in preset</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-priority">--priority</a></kbd></td>
    <td align="LEFT" nowrap>OS/2 process priority control</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#q">-q</a></kbd></td>
    <td align="LEFT" nowrap>algorithm quality selection</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-silent">--quiet</a></kbd></td>
    <td align="LEFT" nowrap>silent operation</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#r">-r</a></kbd></td>
    <td align="LEFT" nowrap>input file is raw PCM</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-replaygain-accurate">--replaygain-accurate</a></kbd></td>
    <td align="LEFT" nowrap>compute ReplayGain more accurately and find the peak sample</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-replaygain-fast">--replaygain-fast</a></kbd></td>
    <td align="LEFT" nowrap>compute ReplayGain fast but slightly inaccurately (default)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-resample">--resample</a></kbd></td>
    <td align="LEFT" nowrap>output sampling frequency in kHz (encoding only)</td>
  </tr>
  <TR VALIGN="TOP"> 
    <TD ALIGN="LEFT" nowrap><kbd><a href="#s">-s</a></kbd></TD>
    <TD ALIGN="LEFT" nowrap>sampling frequency in kHz</TD>
  </TR>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-silent">-S</a></kbd></td>
    <td align="LEFT" nowrap>silent operation</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-scale">--scale</a></kbd></td>
    <td align="LEFT" nowrap>scale input</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-scale-l">--scale-l</a></kbd></td>
    <td align="LEFT" nowrap>scale input channel 0 (left)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-scale-r">--scale-r</a></kbd></td>
    <td align="LEFT" nowrap>scale input channel 1 (right)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-short">--short</a></kbd></td>
    <td align="LEFT" nowrap>use short blocks</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-silent">--silent</a></kbd></td>
    <td align="LEFT" nowrap>silent operation</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-strictly-enforce-ISO">--strictly-enforce-ISO</a></kbd></td>
    <td align="LEFT" nowrap>strict ISO compliance</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#t">-t</a></kbd></td>
    <td align="LEFT" nowrap>disable INFO/WAV header</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#V">-V</a></kbd></td>
    <td align="LEFT" nowrap>VBR quality setting (0...9)</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-vbr-new">--vbr-new</a></kbd></td>
    <td align="LEFT" nowrap>new VBR mode</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-vbr-old">--vbr-old</a></kbd></td>
    <td align="LEFT" nowrap>older VBR mode</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#-verbose">--verbose</a></kbd></td>
    <td align="LEFT" nowrap>verbosity</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#x">-x</a></kbd></td>
    <td align="LEFT" nowrap>swapbytes</td>
  </tr>
  <tr valign="TOP"> 
    <td align="LEFT" nowrap><kbd><a href="#Xquant">-X</a></kbd></td>
    <td align="LEFT" nowrap>change quality measure</td>
  </tr>
</TABLE>
<BR>
<dl> 
  <dt><strong>* <kbd>-a</kbd><a name="a">&nbsp;&nbsp;&nbsp;&nbsp;downmix&#160;</a></strong> 
  <dd>Mix the stereo input file to mono and encode as mono.<br>
    The downmix is calculated as the sum of the left and right channel, attenuated 
    by 6 dB. <br>
    <br>
    This option is only needed in the case of raw PCM stereo input (because LAME 
    cannot determine the number of channels in the input file).<br>
    To encode a stereo PCM input file as mono, use "lame -m s -a".<br>
    <br>
    For WAV and AIFF input files, using "-m m" will always produce a mono .mp3 
    file from both mono and stereo input. 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--abr n</kbd><a name="-abr">&nbsp;&nbsp;&nbsp;&nbsp;average 
    bitrate encoding</a></strong> </dt>
</dl>
<dl> 
  <dd>Turns on encoding with a targeted average bitrate of n kbits, allowing to 
    use frames of different sizes. The allowed range of n is 8-310, you can use 
    any integer value within that range.<br>
    <br>
    It can be combined with the -b and -B switches like:<br>
    lame --abr 123 -b 64 -B 192 a.wav a.mp3<br>
    which would limit the allowed frame sizes between 64 and 192 kbits. <br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--allshort</kbd><a name="-allshort">&nbsp;&nbsp;&nbsp;&nbsp;use 
    short blocks only</a></strong> </dt>
</dl>
<dl> 
  <dd>Use only short blocks, no long ones. 
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--athlower n</kbd><a name="-athlower">&nbsp;&nbsp;&nbsp;&nbsp;lower 
    the ATH</a></strong> </dt>
</dl>
<dl> 
  <dd>Lower the ATH (absolute threshold of hearing) by n dB.<br>
    Normally, humans are unable to hear any sound below this threshold, but for 
    music recorded at very low level this option might be useful.
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--athonly</kbd><a name="-athonly">&nbsp;&nbsp;&nbsp;&nbsp;ATH 
    only</a></strong> </dt>
</dl>
<dl> 
  <dd>This option causes LAME to ignore the output of the psy-model and only use 
    masking from the ATH (absolute threshold of hearing). Might be useful at very 
    high bitrates or for testing the ATH. 
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--athshort</kbd><a name="-athshort">&nbsp;&nbsp;&nbsp;&nbsp;ATH 
    only for short blocks</a></strong> </dt>
</dl>
<dl> 
  <dd>Ignore psychoacoustic model for short blocks, use ATH only. 
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--athtype 0/1/2</kbd><a name="-athtype">&nbsp;&nbsp;&nbsp;&nbsp;select 
    ATH type</a></strong> </dt>
</dl>
<dl> 
  <dd>The Absolute Threshold of Hearing is the minimum threshold under which humans 
    are unable to hear any sound. In the past, LAME was using ATH shape 0 which 
    is the Painter & Spanias formula. Tests have shown that this formula is innacurate 
    for the 13-22 kHz area, leading to audible artifacts in some cases. Shape 1 
    was thus implemented, which is over sensitive, leading to very high bitrates. 
    Shape 2 formula was accurately modelized from real data in order to real optimal 
    quality while not wasting bitrate. In CBR and ABR modes, LAME uses ATH shape 
    2 by default. <br>
    <br>
    In VBR mode, LAME is adapting its shape according to the 
    -V value, going gradually from the 0 shape at -V9 up to shape 2 at -V0.
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>-b n</kbd><a name="b">&nbsp;&nbsp;&nbsp;&nbsp;bitrate</a></strong> 
  </dt>
</dl>
<dl> 
  <dd>For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)<br>
    n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320<br>
    <br>
    For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)<br>
    n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160<br>
    <br>
    Default is 128 kbps for MPEG1 and 64 kbps for MPEG2. <br>
    <br>
    When used with variable bitrate encoding (VBR), -b specifies the minimum bitrate 
    to be used. However, in order to avoid wasted space, the smallest frame size 
    available will be used during silences. 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>-B n</kbd><a name="Bmax">&nbsp;&nbsp;&nbsp;&nbsp;maximum 
    VBR/ABR bitrate&nbsp;</a></strong> </dt>
</dl>
<dl> 
  <dd>For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)<br>
    n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320<br>
    <br>
    For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)<br>
    n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160<br>
    <br>
    Specifies the maximum allowed bitrate when using VBR/ABR <br>
    <br>
    The use of -B is NOT RECOMMENDED. A 128kbps CBR bitstream, because of the bit reservoir, 
    can actually have frames which use as many bits as a 320kbps frame. VBR modes 
    minimize the use of the bit reservoir, and thus need to allow 320kbps frames 
    to get the same flexibility as CBR streams.<br>
    <br>
    <i>note: If you own an mp3 hardware player build upon a MAS 3503 chip, you 
    must set maximum bitrate to no more than 224 kpbs.</i> <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--bitwidth 8/16/24/32</kbd><a name="-bitwidth">&nbsp;&nbsp;&nbsp;&nbsp;input 
    bit width&nbsp;</a></strong> </dt>
</dl>
<dl> 
  <dd> Required only for raw PCM input files. Otherwise it will be determined 
    from the header of the input file. <br>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--clipdetect</kbd><a name="-clipdetect">&nbsp;&nbsp;&nbsp;&nbsp;clipping detection</a></strong> 
  </dt>
</dl>
<dl> 
  <dd>
    Enable --replaygain-accurate and print a message whether clipping 
    occurs and how far in dB the waveform is from full scale.<br>
    <br>
    This option is not usable if the MP3 decoder was <b>explicitly</b>
    disabled in the build of LAME.<br>
    <br>
    See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>
  <dt><br>
    <br>
    <hr width="50%" noshade align="center">
    <br>
  <dt><strong>* <kbd>--cbr</kbd><a name="-cbr">
    &nbsp;&nbsp;&nbsp;&nbsp;enforce use of constant bitrate</a></strong> 
  </dt>
</dl>
<dl> 
  <dd>This switch enforces the use of constant bitrate encoding. 
  <dt><br>
    <br>
    <hr width="50%" noshade align="center">
    <br>
  <dt><strong>* <kbd>--cbr</kbd><a name="-cbr">
    &nbsp;&nbsp;&nbsp;&nbsp;enforce use of constant bitrate</a></strong> 
  </dt>
</dl>
<dl> 
  <dd>This switch enforces the use of constant bitrate encoding. 
  <dt><br>
    <br>
    <hr width="50%" noshade align="center">
    <br>
  <dt><strong>* <kbd>--comp</kbd><a name="-comp">&nbsp;&nbsp;&nbsp;&nbsp;choose 
    compression ratio</a></strong> </dt>
</dl>
<dl> 
  <dd>Instead of choosing bitrate, using this option, user can choose compression 
    ratio to achieve. 
  <dt><br>
    <br>
    <hr width="50%" noshade align="center">
    <br>
  <dt><strong>* <kbd>--cwlimit n</kbd><a name="-cwlimit">&nbsp;&nbsp;&nbsp;tonality 
    limit</a></strong> </dt>
</dl>
<dl> 
  <dd>Compute tonality up to freq (in kHz). Default setting is 8.8717. 
  <dt><br>
    <br>
    <hr width="50%" noshade align="center">
    <br>
  <dt><strong>* <kbd>-d</kbd><a name="d">&nbsp;&nbsp;&nbsp;&nbsp;block type control</a></strong> 
  </dt>
</dl>
<dl> 
  <dd>Allows the left and right channels to use different block size types. 
  <dt><br>
    <br>
    <hr width="50%" noshade align="center">
    <br>
  <dt><strong>* <kbd>--decode</kbd><a name="-decode">&nbsp;&nbsp;&nbsp;&nbsp;decoding 
    only</a></strong> </dt>
</dl>
<dl> 
  <dd>Uses LAME for decoding to a WAV file. The input file can be any input type 
    supported by encoding, including layer I,II,III (MP3) and OGG files. In case 
    of MPEG files, LAME uses a bugfixed version of mpglib for decoding.<br>
    <br>
    If -t is used (disable WAV header), Lame will output raw PCM in native endian 
    format. You can use -x to swap bytes order. <br>
    <br>
    This option is not usable if the MP3 decoder was <b>explicitly</b>
    disabled in the build of LAME.
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--disptime n</kbd><a name="-disptime">&nbsp;&nbsp;&nbsp;&nbsp;time 
    between display updates</a></strong> </dt>
</dl>
<dl> 
  <dd>Set the delay in seconds between two display updates. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-e n/5/c</kbd><a name="e">&nbsp;&nbsp;&nbsp;&nbsp;de-emphasis</a></strong> 
  </dt>
</dl>
<dl> 
  <dd> <br>
    n = (none, default)<br>
    5 = 0/15 microseconds<br>
    c = citt j.17<br>
    <br>
    All this does is set a flag in the bitstream. If you have a PCM input file 
    where one of the above types of (obsolete) emphasis has been applied, you 
    can set this flag in LAME. Then the mp3 decoder should de-emphasize the output 
    during playback, although most decoders ignore this flag.<br>
    <br>
    A better solution would be to apply the de-emphasis with a standalone utility 
    before encoding, and then encode without -e. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-f</kbd><a name="f">&nbsp;&nbsp;&nbsp;&nbsp;fast mode</a></strong> 
  </dt>
</dl>
<dl> 
  <dd> This switch forces the encoder to use a faster encoding mode, but with 
    a lower quality. The behaviour is the same as the -q7 switch.<br>
    <br>
    Noise shaping will be disabled, but psycho acoustics will still be computed 
    for bit allocation and pre-echo detection. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-F</kbd><a name="FF">&nbsp;&nbsp;&nbsp;strictly enforce the 
    -b option</a></strong> </dt>
</dl>
<dl> 
  <dd> This is mainly for use with hardware players that do not support low bitrate 
    mp3.<br>
    <br>
    Without this option, the minimum bitrate will be ignored for passages of analog 
    silence, ie when the music level is below the absolute threshold of human 
    hearing (ATH). 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--freeformat</kbd><a name="-freeformat">&nbsp;&nbsp;&nbsp;&nbsp;free 
    format bitstream</a></strong> </dt>
</dl>
<dl> 
  <dd> Produces a free format bitstream. With this option, you can use -b with 
    any bitrate higher than 8 kbps.<br>
    <br>
    However, even if an mp3 decoder is required to support free bitrates at least 
    up to 320 kbps, many players are unable to deal with it.<br>
    <br>
    Tests have shown that the following decoders support free format:<br>
    <br>
    FreeAmp up to 440 kbps<br>
    in_mpg123 up to 560 kbps<br>
    l3dec up to 310 kbps<br>
    LAME up to 560 kbps<br>
    MAD up to 640 kbps<br>
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-h</kbd><a name="h">&nbsp;&nbsp;&nbsp;&nbsp;high quality</a></strong> 
  </dt>
</dl>
<dl> 
  <dd> Use some quality improvements. Encoding will be slower, but the result 
    will be of higher quality. The behaviour is the same as the -q2 switch.<br>
    This switch is always enabled when using VBR. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--help</kbd><a name="-help">&nbsp;&nbsp;&nbsp;&nbsp;help</a></strong> 
  </dt>
</dl>
<dl> 
  <dd> Display a list of all available options. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--highpass</kbd><a name="-highpass">&nbsp;&nbsp;&nbsp;&nbsp;highpass 
    filtering frequency in kHz</a></strong> </dt>
</dl>
<dl> 
  <dd> Set an highpass filtering frequency. Frequencies below the specified one 
    will be cutoff. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--highpass-width</kbd><a name="-highpass-width">&nbsp;&nbsp;&nbsp;&nbsp;width 
    of highpass filtering in kHz</a></strong> </dt>
</dl>
<dl> 
  <dd> Set the width of the highpass filter. The default value is 15% of the highpass 
    frequency. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-k</kbd><a name="k">&nbsp;&nbsp;&nbsp;&nbsp;full bandwidth</a></strong> 
  </dt>
</dl>
<dl> 
  <dd> Tells the encoder to use full bandwidth and to disable all filters. By 
    default, the encoder uses some lowpass filtering at lower bitrates, in order 
    to keep a good quality by giving more bits to more important frequencies.<br>
    Increasing the bandwidth from the default setting might produce ringing artefacts 
    at low bitrates. Use with care! 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--lowpass</kbd><a name="-lowpass">&nbsp;&nbsp;&nbsp;&nbsp;lowpass 
    filtering frequency in kHz</a></strong></dt>
</dl>
<dl> 
  <dd> Set a lowpass filtering frequency. Frequencies above the specified one 
    will be cutoff. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--lowpass-width</kbd><a name="-lowpass-width">&nbsp;&nbsp;&nbsp;&nbsp;width 
    of lowpass filtering in kHz</a></strong></dt>
</dl>
<dl> 
  <dd> Set the width of the lowpass filter. The default value is 15% of the lowpass 
    frequency. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-m s/<b>j/</b>f/d/m</kbd><a name="m">&nbsp;&nbsp;&nbsp;&nbsp;stereo 
    mode</a></strong> </dt>
</dl>
<dl> 
  <dd> Joint-stereo is the default mode for input files featuring two channels.. 
    <b><i><br>
    <br>
    stereo</i></b> <br>
    In this mode, the encoder makes no use of potentially existing correlations 
    between the two input channels. It can, however, negotiate the bit demand 
    between both channel, i.e. give one channel more bits if the other contains 
    silence or needs less bits because of a lower complexity.<br>
    <br>
    <i><b>joint stereo</b></i><br>
    In this mode, the encoder will make use of correlation between both channels. 
    The signal will be matrixed into a sum ("mid"), computed by L+R, and difference 
    ("side") signal, computed by L-R, and more bits are allocated to the mid channel.<br>
    This will effectively increase the bandwidth if the signal does not have too 
    much stereo separation, thus giving a significant gain in encoding quality. 
    In joint stereo, the encoder can select between Left/Right and Mid/Side representation 
    on a frame basis.<br>
    <br>
    Using mid/side stereo inappropriately can result in audible compression artifacts. 
    To much switching between mid/side and regular stereo can also sound bad. 
    To determine when to switch to mid/side stereo, LAME uses a much more sophisticated 
    algorithm than that described in the ISO documentation, and thus is safe to 
    use in joint stereo mode.<br>
    <br>
    <b><i>forced joint stereo </i></b><br>
    This mode will force MS joint stereo on all frames. It's slightly faster than 
    joint stereo, but it should be used only if you are sure that every frame 
    of the input file has very little stereo separation.<br>
    <br>
    <b><i>dual channels </i></b><br>
    In this mode, the 2 channels will be totally independently encoded. Each 
    channel will have exactly half of the bitrate. This mode is designed for applications 
    like dual languages encoding (ex: English in one channel and French in the 
    other). Using this encoding mode for regular stereo files will result in a 
    lower quality encoding.<br>
    <br>
    <b><i>mono</i></b><br>
    The input will be encoded as a mono signal. If it was a stereo signal, it 
    will be downsampled to mono. The downmix is calculated as the sum of the left 
    and right channel, attenuated by 6 dB. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--mp1input</kbd><a name="-mp1input">&nbsp;&nbsp;&nbsp;&nbsp;MPEG 
    Layer I input file</a></strong> </dt>
</dl>
<dl> 
  <dd> Assume the input file is a MPEG Layer I file.<br>
    If the filename ends in ".mp1" or &quot;.mpg&quot; LAME will assume it is 
    a MPEG Layer I file. For stdin or Layer I files which do not end in .mp1 or .mpg 
    you need to use this switch. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--mp2input</kbd><a name="-mp2input">&nbsp;&nbsp;&nbsp;&nbsp;MPEG 
    Layer II input file</a></strong> </dt>
</dl>
<dl> 
  <dd> Assume the input file is a MPEG Layer II (ie MP2) file.<br>
    If the filename ends in ".mp2" LAME will assume it is a MPEG Layer II file. For 
    stdin or Layer II files which do not end in .mp2 you need to use this switch. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--mp3input</kbd><a name="-mp3input">&nbsp;&nbsp;&nbsp;&nbsp;MPEG 
    Layer III input file</a></strong> </dt>
</dl>
<dl> 
  <dd> Assume the input file is a MP3 file. Useful for downsampling from one 
    mp3 to another. As an example, it can be useful for streaming through an 
    IceCast server.<br>
    If the filename ends in ".mp3" LAME will assume it is an MP3 file. For stdin or 
    MP3 files which do not end in .mp3 you need to use this switch. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--noath</kbd><a name="-noath">&nbsp;&nbsp;&nbsp;&nbsp;disable 
    ATH</a></strong> </dt>
</dl>
<dl> 
  <dd> Disable any use of the ATH (absolute threshold of hearing) for masking. 
    Normally, humans are unable to hear any sound below this threshold. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--noasm mmx/3dnow/sse</kbd><a name="-noasm">
   &nbsp;&nbsp;&nbsp;&nbsp;disable assembly optimizations</a></strong> </dt>
</dl>
<dl> 
  <dd>Disable specific assembly optimizations. Quality will not increase, only
      speed will be reduced. If you have problems running Lame on a Cyrix/Via
      processor, disabling mmx optimizations might solve your problem.
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--nohist</kbd><a name="-nohist">&nbsp;&nbsp;&nbsp;&nbsp;disable 
    histogram display</a></strong> </dt>
</dl>
<dl> 
  <dd> By default, LAME will display a bitrate histogram while producing VBR mp3 
    files. This will disable that feature.<br>
    Histogram display might not be available on your release. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--noreplaygain</kbd><a name="-noreplaygain">&nbsp;&nbsp;&nbsp;&nbsp;disable 
    ReplayGain analysis</a></strong></dt>
</dl>
<dl> 
  <dd> By default ReplayGain analysis is enabled. This switch disables it.<br>
  <br>
    See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>,
    <a href="#-replaygain-fast">--replaygain-fast</a>
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--nores</kbd><a name="-nores">&nbsp;&nbsp;&nbsp;&nbsp;disable 
    bit reservoir</a></strong></dt>
</dl>
<dl> 
  <dd> Disable the bit reservoir. Each frame will then become independent from 
    previous ones, but the quality will be lower. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--noshort</kbd><a name="-noshort">&nbsp;&nbsp;&nbsp;&nbsp;disable 
    short blocks frames</a></strong></dt>
</dl>
<dl> 
  <dd> Encode all frames using long blocks only. This could increase quality when 
    encoding at very low bitrates, but can produce serious pre-echo artefacts. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--notemp</kbd><a name="-notemp">&nbsp;&nbsp;&nbsp;&nbsp;disable 
    temporal masking</a></strong></dt>
</dl>
<dl> 
  <dd>Don't make use of the temporal masking effect. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-o</kbd><a name="o">&nbsp;&nbsp;&nbsp;&nbsp;non-original</a></strong> 
  </dt>
</dl>
<dl> 
  <dd> Mark the encoded file as being a copy. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-p</kbd><a name="p">&nbsp;&nbsp;&nbsp;&nbsp;error protection</a></strong></dt>
</dl>
<dl> 
  <dd> Turn on CRC error protection.<br>
    It will add a cyclic redundancy check (CRC) code in each frame, allowing to 
    detect transmission errors that could occur on the MP3 stream. However, it 
    takes 16 bits per frame that would otherwise be used for encoding, and then 
    will slightly reduce the sound quality. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--preset presetName</kbd> <a name="-preset">&nbsp;&nbsp;&nbsp;&nbsp;use 
    built-in preset</a></strong></dt>
</dl>
<dd> Use one of the built-in presets (standard, fast standard, extreme, fast extreme, insane, or the abr/cbr modes).
<br>
<dd>   "--preset help" gives more information about the usage possibilities for these presets. 
<dt><br>
  <br>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--priority 0...4</kbd><a name="-priority">&nbsp;&nbsp;&nbsp;&nbsp;OS/2 
  process priority control</a></strong> </dt>
<dl> 
  <dd> With this option, LAME will run with a different process priority under 
    IBM OS/2.<br>
    This will greatly improve system responsiveness, since OS/2 will have more 
    free time to properly update the screen and poll the keyboard/mouse. It should 
    make quite a difference overall, especially on slower machines. LAME's performance 
    impact should be minimal.<br>
    <br>
  <dd><b>0 (Low priority)</b><br>
    Priority 0 assumes "IDLE" class, with delta 0.<br>
    LAME will have the lowest priority possible, and the encoding may be suspended 
    very frequently by user interaction.<br>
    <br>
  <dd><b>1 (Medium priority)</b><br>
    Priority 1 assumes "IDLE" class, with delta +31.<br>
    LAME won't interfere at all with what you're doing.<br>
    Recommended if you have a slower machine. <br>
    <br>
  <dd><b>2 (Regular priority)</b><br>
    Priority 2 assumes "REGULAR" class, with delta -31.<br>
    LAME won't interfere with your activity. It'll run just like a regular process, 
    but will spare just a bit of idle time for the system. Recommended for most 
    users. <br>
    <br>
  <dd><b>3 (High priority)</b><br>
    Priority 3 assumes "REGULAR" class, with delta 0.<br>
    LAME will run with a priority a bit higher than a normal process. <br>
    Good if you're just running LAME by itself or with moderate user interaction.<br>
    <br>
  <dd><b>4 (Maximum priority)</b><br>
    Priority 4 assumes "REGULAR" class, with delta +31.<br>
    LAME will run with a very high priority, and may interfere with the machine 
    response.<br>
    Recommended if you only intend to run LAME by itself, or if you have a fast 
    processor. <br>
    <br>
    <br>
    Priority 1 or 2 is recommended for most users. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-q 0..9</kbd><a name="q">&nbsp;&nbsp;&nbsp;&nbsp;algorithm 
    quality selection</a></strong></dt>
</dl>
<dl> 
  <dd> Bitrate is of course the main influence on quality. The higher the bitrate, 
    the higher the quality. But for a given bitrate, we have a choice of algorithms 
    to determine the best scalefactors and Huffman encoding (noise shaping).<br>
    <br>
    -q 0: use slowest &amp; best possible version of all algorithms. -q 0 and -q 1 
    are slow and may not produce significantly higher quality.<br>
    <br>
    -q 2: recommended. Same as -h.<br>
    <br>
    -q 5: default value. Good speed, reasonable quality.<br>
    <br>
    -q 7: same as -f. Very fast, ok quality. (psycho acoustics are used for pre-echo 
    &amp; M/S, but no noise shaping is done.<br>
    <br>
    -q 9: disables almost all algorithms including psy-model. poor quality. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-r</kbd><a name="r">&nbsp;&nbsp;&nbsp;&nbsp;input file is 
    raw PCM</a></strong></dt>
</dl>
<dl> 
  <dd> Assume the input file is raw PCM. Sampling rate and mono/stereo/jstereo 
    must be specified on the command line. Without -r, LAME will perform several 
    fseek()'s on the input file looking for WAV and AIFF headers.<br>
    Might not be available on your release. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--replaygain-accurate</kbd><a name="-replaygain-accurate">&nbsp;&nbsp;&nbsp;&nbsp;compute
   ReplayGain more accurately and find the peak sample</a></strong></dt>
</dl>
<dl> 
  <dd>
    Enable decoding on the fly. Compute "Radio" ReplayGain on the decoded 
    data stream. Find the peak sample of the decoded data stream and store 
    it in the file.<br>
    <br>    
    ReplayGain analysis does <i>not</i> affect the content of a 
    compressed data stream itself, it is a value stored in the header 
    of a sound file. Information on the purpose of ReplayGain and the
    algorithms used is available from 
    <a href="http://www.replaygain.org/">http://www.replaygain.org/</a><br>
    <br>
    By default, LAME performs ReplayGain analysis on the input data 
    (after the user-specified volume scaling). This
    behavior might give slightly inaccurate results because the data on 
    the output of a lossy compression/decompression sequence differs from 
    the initial input data. When --replaygain-accurate is specified the
    mp3 stream gets decoded on the fly and the analysis is performed on the
    decoded data stream. Although theoretically this method gives more 
    accurate results, it has several disadvantages:
    <ul>
      <li> tests have shown that the difference between the ReplayGain values 
        computed on the input data and decoded data is usually no greater 
        than 0.5dB, although the minimum volume difference the human ear 
        can perceive is about 1.0dB
      </li>
      <li> decoding on the fly significantly slows down the encoding process
      </li>
    </ul>
    The apparent advantage is that:
    <ul>
      <li> with --replaygain-accurate the peak sample is determined and 
        stored in the file. The knowledge of the peak sample can be useful
        to decoders (players) to prevent a negative effect called 'clipping'
        that introduces distortion into sound.
      </li>
    </ul>    
    <br>
    Only the "RadioGain" ReplayGain value is computed. It is stored in the 
    LAME tag. The analysis is  performed with the reference volume equal
    to 89dB. Note: the reference volume has been changed from 83dB on 
    transition from version 3.95 to 3.95.1.<br>
    <br>
    This option is not usable if the MP3 decoder was <b>explicitly</b>
    disabled in the build of LAME. (Note: if LAME is compiled without the 
    MP3 decoder, ReplayGain analysis is performed on the input data after
    user-specified volume scaling).<br>
    <br>
    See also: <a href="#-replaygain-fast">--replaygain-fast</a>, 
    <a href="#-noreplaygain">--noreplaygain</a>, <a href="#-clipdetect">--clipdetect</a>
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--replaygain-fast</kbd><a name="-replaygain-fast">&nbsp;&nbsp;&nbsp;&nbsp;compute
   ReplayGain fast but slightly inaccurately (default)</a></strong></dt>
</dl>
<dl> 
  <dd>
    Compute "Radio" ReplayGain on the input data stream after user-specified 
    volume scaling and/or resampling.<br>
    <br>    
    ReplayGain analysis does <i>not</i> affect the content of a 
    compressed data stream itself, it is a value stored in the header 
    of a sound file. Information on the purpose of ReplayGain and the
    algorithms used is available from 
    <a href="http://www.replaygain.org/">http://www.replaygain.org/</a><br>
    <br>
    Only the "RadioGain" ReplayGain value is computed. It is stored in the 
    LAME tag. The analysis is  performed with the reference volume equal
    to 89dB. Note: the reference volume has been changed from 83dB on 
    transition from version 3.95 to 3.95.1.<br>
    <br>
    This switch is enabled by default.<br>
    <br>
    See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>, 
    <a href="#-noreplaygain">--noreplaygain</a>
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--resample 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="-resample">&nbsp;&nbsp;&nbsp;&nbsp;output 
    sampling frequency in kHz</a></strong></dt>
</dl>
<dl> 
  <dd> Select output sampling frequency (for encoding only). <br>
    If not specified, LAME will automatically resample the input when using high 
    compression ratios. 
  <dt><br>
  </dt>
</dl>
<dl> 
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-s 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="s">&nbsp;&nbsp;&nbsp;&nbsp;sampling 
    frequency</a></strong> </dt>
</dl>
<dl> 
  <dd> Required only for raw PCM input files. Otherwise it will be determined 
    from the header of the input file.<br>
    <br>
    LAME will automatically resample the input file to one of the supported MP3 
    samplerates if necessary. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-S / --silent / --quiet</kbd><a name="-silent">&nbsp;&nbsp;&nbsp;&nbsp;silent 
    operation</a></strong> </dt>
</dl>
<dl> 
  <dd> Don't print progress report. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--scale n</kbd><a name="-scale">&nbsp;&nbsp;&nbsp;&nbsp;scales 
    input by n</a></strong> </dt>
  <dt><strong>* <kbd>--scale-l n</kbd><a name="-scale-l">&nbsp;&nbsp;&nbsp;&nbsp;scales 
    input channel 0 (left) by n</a></strong> </dt>
  <dt><strong>* <kbd>--scale-r n</kbd><a name="-scale-r">&nbsp;&nbsp;&nbsp;&nbsp;scales 
    input channel 1 (right) by n</a></strong> </dt>
</dl>
<dl> 
  <dd>Scales input by n. This just multiplies the PCM data (after it has been 
    converted to floating point) by n. <br>
    <br>
    n > 1: increase volume<br>
    n = 1: no effect<br>
    n < 1: reduce volume<br>
    <br>
    Use with care, since most MP3 decoders will truncate data which decodes to 
    values greater than 32768. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--short</kbd><a name="-short">&nbsp;&nbsp;&nbsp;&nbsp;use 
    short blocks</a></strong> </dt>
</dl>
<dl> 
  <dd>Let LAME use short blocks when appropriate. It is the default setting. 
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--strictly-enforce-ISO</kbd><a name="-strictly-enforce-ISO">&nbsp;&nbsp;&nbsp;&nbsp;strict 
    ISO compliance</a></strong> </dt>
</dl>
<dl> 
  <dd> With this option, LAME will enforce the 7680 bit limitation on total frame 
    size.<br>
    This results in many wasted bits for high bitrate encodings but will ensure 
    strict ISO compatibility. This compatibility might be important for hardware 
    players. 
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-t</kbd><a name="t">&nbsp;&nbsp;&nbsp;&nbsp;disable INFO/WAV 
    header </a></strong></dt>
</dl>
<dl> 
  <dd> Disable writing of the INFO Tag on encoding.<br>
    This tag in embedded in frame 0 of the MP3 file. It includes some information 
    about the encoding options of the file, and in VBR it lets VBR aware players 
    correctly seek and compute playing times of VBR files.<br>
    <br>
    When '--decode' is specified (decode to WAV), this flag will disable writing 
    of the WAV header. The output will be raw PCM, native endian format. Use -x 
    to swap bytes. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-V 0...9</kbd><a name="V">&nbsp;&nbsp;&nbsp;&nbsp;VBR quality 
    setting</a></strong></dt>
</dl>
<dl> 
  <dd> Enable VBR (Variable BitRate) and specifies the value of VBR quality.<br>
    default=4<br>
    0=highest quality. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--vbr-new</kbd><a name="-vbr-new">&nbsp;&nbsp;&nbsp;&nbsp;new 
    VBR mode</a></strong></dt>
</dl>
<dl> 
  <dd> Invokes the newest VBR algorithm. During the development of version 3.90, 
    considerable tuning was done on this algorithm, and it is now considered to 
    be on par with the original --vbr-old. <br>
    It has the added advantage of being very fast (over twice as fast as --vbr-old). 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--vbr-old</kbd><a name="-vbr-old">&nbsp;&nbsp;&nbsp;&nbsp;older 
    VBR mode</a></strong></dt>
</dl>
<dl> 
  <dd> Invokes the oldest, most tested VBR algorithm. It produces very good quality 
    files, though is not very fast. This has, up through v3.89, been considered 
    the "workhorse" VBR algorithm. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--verbose</kbd><a name="-verbose">&nbsp;&nbsp;&nbsp;&nbsp;verbosity</a></strong></dt>
</dl>
<dl> 
  <dd> Print a lot of information on screen. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-x</kbd><a name="x">&nbsp;&nbsp;&nbsp;&nbsp;swapbytes</a></strong> 
  </dt>
</dl>
<dl> 
  <dd> Swap bytes in the input file or output file when using --decode.<br>
    For sorting out little endian/big endian type problems. If your encodings 
    sounds like static, try this first. 
  <dt><br>
    <br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-X 0...7</kbd><a name="Xquant">&nbsp;&nbsp;&nbsp;&nbsp;change 
    quality measure</a></strong> </dt>
</dl>
<dl> 
  <dd> When LAME searches for a "good" quantization, it has to compare the actual 
    one with the best one found so far. The comparison says which one is better, 
    the best so far or the actual. The -X parameter selects between different 
    approaches to make this decision, -X0 being the default mode:<br>
    <br>
    <b>-X0 </b><br>
    The criterions are (in order of importance):<br>
    * less distorted scalefactor bands<br>
    * the sum of noise over the thresholds is lower<br>
    * the total noise is lower<br>
    <br>
    <b>-X1</b><br>
    The actual is better if the maximum noise over all scalefactor bands is less 
    than the best so far .<br>
    <br>
    <b>-X2</b><br>
    The actual is better if the total sum of noise is lower than the best so far.<br>
    <br>
    <b>-X3</b><br>
    The actual is better if the total sum of noise is lower than the best so far 
    and the maximum noise over all scalefactor bands is less than the best so 
    far plus 2db.<br>
    <br>
    <b>-X4</b> <br>
    Not yet documented.<br>
    <br>
    <b>-X5</b><br>
    The criterions are (in order of importance):<br>
    * the sum of noise over the thresholds is lower <br>
    * the total sum of noise is lower<br>
    <br>
    <b>-X6</b> <br>
    The criterions are (in order of importance):<br>
    * the sum of noise over the thresholds is lower<br>
    * the maximum noise over all scalefactor bands is lower<br>
    * the total sum of noise is lower<br>
    <br>
    <b>-X7</b> <br>
    The criterions are:<br>
    * less distorted scalefactor bands<br>
    or<br>
    * the sum of noise over the thresholds is lower 
</dl>
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